Convert audio files for Asterisk
Convert audio files for Asterisk
Converting to sln format
Starting from Asterisk 1.2.0, the .sln (SLINEAR) format seems to be the preferred format.
To convert wav file to sln, use the following command:
sox foo-in.wav -t raw -r 8000 -s -w -c 1 foo-out.sln
Note that sox v14.3.0 and above (installed in Ubuntu 9.10), the -w option has changed to -2
sox foo-in.wav -t raw -r 8000 -s -2 -c 1 foo-out.sln
If you have a directory full of .wav files to convert, try this command. It uses sed to automatically rename the files with the .sln extension (assuming incoming wav files at a sample rate other than 8khz.)
for a in *.wav;
do sox “$a” -t raw -r 8000 -s -w -c 1 `echo $a|sed “s/.wav/.sln/”` resample -ql;
done
Converting your WAV files to good GSM files is easier than you might think if you have the program Sox installed (on Debian systems the libsox-fmt-gsm package is required in addition to sox). From the shell prompt, enter this command:
sox foo.wav -r 8000 foo.gsm resample -ql
and hit the <ENTER> key. Note that the sox option ‘-ql’ (lower case L) modifies the resample option. It is not a number one (1).
In a few moments you will have a new GSM format file in the same directory as the original WAV file.
In this example “foo.wav” is your main voice menu audio file in WAV format, and “foo.gsm” is the same file converted to GSM format.
If you wanted to, you could use “main-voice-menu.gsm” as the name in place of “foo.gsm”: what matters here is the second file name you use in this command ends in “.gsm”.
If your WAV file was in stereo, add the -c1 option to convert to mono, or the output will sound very strange.
sox foo.wav -r 8000 -c1 foo.gsm resample -ql
You may get better results if you record your WAV file in 16 bit 8000 Hz mono and then run
sox foo.wav foo.gsm
If you have multiple WAV files in one directory and you want to convert them all, use this command:
for a in *.wav; do sox “$a” -r 8000 -c1 “`echo $a|sed -e s/wav//`gsm” resample -ql; done
You can also put a bash script in /usr/bin and name it wav-gsm-convert. The content can be like this
- !/bin/bash
s=`echo $1| sed -e’s/\.wav//’|xargs -i{} echo {}.gsm`
sox -t wav $1 -r 8000 -c1 -t gsm $s resample -ql
Next, move your new foo.gsm file to the directory: /var/lib/asterisk/sounds
Now you can easily use the applications Playback and Background in your extensions.conf file to play your fabulous main voice menu. For example:
exten => s,1,Background(foo)
or
exten => s,1,Background(main-voice-menu)
or
exten => s,1,Playback(foo)
or
exten => s,1,Playback(main-voice-menu)
Playing .sln files from the command line
You can play sln files using sox from the command line (play is part of sox):
play -t raw -r 8000 -s -w -c 1 file.sln
Using WAV files
Asterisk has codecs for wav (pcm), gsm, g729, g726, and wav49, all of which can be used for Playback and Background. However, Asterisk does not understand ADPCM WAV files.
To convert your WAV files to a format which Asterisk can understand, use the following command:
sox foo-in.wav -r 8000 -c 1 -s -w foo-out.wav resample -ql
Note that sox v14.3.0 and above (installed in Ubuntu 9.10), resample is no longer used, remix is used or leave it as:
sox foo-in.wav -r 8k -c 1 -s -w foo-out.wav
All so if your using sox v14.3.foo and above and you are getting check_header errors with play back try this:
sox foo-in.wav -r 8k -c 1 -e gsm foo-out.wav
This also works converting mp3, just make sure you have libsox-fmt-mp3 installed.
# Minimal conversion example sox input.mp3 output.wav # Convert to mono (two possibilities: by specifying output format # or with the 'channels' effect. sox input.mp3 -c 1 output.wav sox input.mp3 output.wav channels 1 # Change sample rate (again two possibilities) sox input.mp3 -r 8000 output.wav sox input.mp3 output.wav rate 8000 # Newer versions of SoX also support sox input.mp3 output.wav rate 8k # Trim a fragment of 30 seconds at an offset of 60 seconds # with the 'trim' effect sox input.mp3 output.wav trim 60 30 # All together now (trimmed fragment in mono, 22.05 Hz sample rate) sox input.mp3 output.wav trim 60 30 channels 1 rate 22050
Normalizing volume and reducing volume fluxuations
Using the sox command’s “compand” filter, you can reduce or eliminate flutters in volume level, and you can normalize the volume of your sound files. This is called dynamic range compression. The effect is desirable for a PBX system where changes in volume, or voices that are too quiet to understand would be considered unprofessional. It is important to perform the compand effect BEFORE you resample it as to preserve as much quality as possible.
An example command to perform some appropriate dynamic range compression and normalization is shown below:
sox “foo-in.wav” -r 8000 -c1 “foo-out.gsm” lowpass 4000 compand 0.02,0.05 -60,-60,-30,-10,-20,-8,-5,-8,-2,-8 -8 -7 0.05 resample -ql
In this example, foo-in.wav is a 16-bit mono 44khz uncompressed pcm WAV file.
Converting to a CD writable format for normal use
So, you’ve decided to do your call recording in GSM format as you don’t care about quality and you don’t want to stuff your disks full, but how do you write that file to an audio CD to send to somebody who wants to listen to the call?
sox infile.gsm -r 44100 -a outfile.wav
